similar to: Play audio into meetme conference?

The subject pretty much says it all. I have a customer who would like to have an option where a caller can leave a voicemail in such a fashion that it would be simultaneously delivered to a set of mailboxes all at once--the idea is "trouble ticket" type operation where multiple technicians will *each* get the vm. He prefers that, if we can do it, to a "shared mailbox"

I've looked in the obvious places but haven't found a definitive answer to the following: can an IAX extension (an Iaxy phone, for instance) do call pickup via *8? Adolfo

Hi everybody, I would like to create SIP call flow Diagram under Windows. Is anybody know a program to perform it? I have already Ethereal and I would like an explicit diagram just to show where something have problems... Thanks Ignace

Hia .... i try to use the fonction Getvar of asterisk to get a variable myDNIS that i have define. i use it as follow Action: Getvar Channel: SIP... Variable: myDNIS but asterisk don't know it .i have the response as follow Response: Error Message: Invalid/unknown command does everybody meet this problem . i try all possible combination and nothing help please ..!! :-( thanks in advance

Hi. I was wondering how can I make incoming calls to wait if the phone operator is busy. I've 8 incoming lines, with 30 extensions. What I need is if the operator is busy with call nr #1 , the new incoming call waits until the op. is free. Looking into app_queue seems the way to go. So I want to ask if I'm right or wrong: I set up only a queue , is to say operatorq, where the only member

Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress

Hello, For those interested in overcoming the problem with some NATs and Firewalls in regards to tftp. I found a nice little tftp server here: http://freshmeat.net/projects/jtftp/?topic_id=87 I tried it and it works great. Regards, Andres. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi. Has anyone experienced with the new firmware .77 ? There's Day Light Saving time now, but haven't time to play with it, till now. Matteo. -- Matteo Brancaleoni Espia System Administrator - IT services Website : http://www.espia.it Email : mbrancaleoni@espia.it

<P>hi,</P> <P>Can anybody pls tell me how to activate loggin CDR on mysql db. I tried&nbsp;editing the /etc/asterisk/cdr_mysql.conf file and recompiling asterisk, it didn't work. Normal loggin on Master.csv file works fine.</P> <P>Thank you inadvance,</P> <P>Surajee</P> <P>&nbsp;</P><br> --------------This mail sent

Hi 2 all. I was thinking to start a little web site with phpwiki, to let the * community build a sort of shared documentation 'bout * & related. That because in a wiki "place" all grows faster, and is also the right place to share experiences. For example it's right to have documentation about * installations, ie who has done what with asterisk Till now we don't know

Hi. I was wondering if there's a parameter to limit the number of concurrent sessions in IAX, globally or on a per-user basis. That could be needed for security purposes (to prevent dos attacks), to limit bandwidth / cpu usage, or to not allow more than N guest connections, for example. Any other VoIP channel support that? (like SIP, MGCP) Matteo. -- Brancaleoni Matteo

Hi. I noticed that if new messages are recorded with voicemail2 , they're not detected by the message waiting indicator, so the mailbox=XXXX param has no effect, and no message waiting is sent to the phone (sip & adsi, or stutter dialtone) Any hint? -- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmgi Srl

Please, don't hate me because I use Redhat. I am aware that I am asking for problems in running Asterisk on Redhat. I recently aquired a nifty server, moved my digium cards, and installed asterisk. I noticed that one of the four processors was being used at 100% and nothing was working. I tracked CPU utilization back to the Asterisk process. Please, help. James

Hello guys, I have searched high and low, but not found any information about rules of using DTMF in SIP INFO method. Cisco has described something with Signal=, but it look like this feature is dependent on implementors? The problem is chan_sip.c cannot correctly translate received DTMF digits, especially #,*. At least with my Antek EGW-804 gateway. Looking into chan_sip.c, I found this code:

I was wondering if the accountcode flag works with sip channels. I was looking into the debug and ,even if I have the line accountcode=XXX into the users sections of my sip.conf, I don't see it logged into the cdr. Matteo Brancaleoni mbrancaleoni@espia.it Emmegi System Administrator EspiA - EMMEGI Srl - e*solution provider Uffici: Via Pascoli, 37 20129 Milano - Italy Sede Legale: Corso

I was wondering what are the values for sending dmtf via sip info. I mean, when I use dtmf relay via sip info, the sip/sdp message contains a Signal=X where X is the dmtf. That's ok for dtmf 0-9 . but what when dtmf is * or # ? we must send signal=# ? I ask that because I noticed that budgetones phone sends out * as signal=10 and # as signal=11 . but asterisk don't detect them, 'cause

Hi. Since a customer requested us that feature, I wrote this little patch for app_dial to allow to play an announcement to the called party, as soon he answers. you can define the file to play in the dial() option, using A(filename). for example: exten => blah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt) that doesn't break anything ... feel free to blame me for anything bad this patch

Hi, All Is there a provision for "AbsoluteTimeout" application to notify called and calling party of the reason why the call suddenly ended? This way, the parties will be much better informed, hence they will/should not think that their VOIP/telco provider(s) are providing bad service. Ta SJ

We have officially made the first release candidate of Zaptel, Libpri, Asterisk and Gastman available. While there are still open major bugs, they are relatively limited, and it was time to go ahead and get the 1.0 ball rolling in earnest. ftp://ftp.digium.com/pub/asterisk Enjoy the code. Special thanks to all the bug marshals and contributers and to everyone who has supported Asterisk

I wish 'A(x)' was available with AgentCallBackLogin!! :( -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Matteo Brancaleoni Sent: Tuesday, February 03, 2004 6:48 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Playing announcement to called user prior toConfirmation show application